Microphone Arrays

Musical instruments create a spatial sound impression. This chapter provides an introduction to the acoustics of sound propagation from musical instruments. An overview of microphone array techniques to measure the sound radiation characteristics from the near and the far field is given. The complex point source model simplifies the physical constellation and describes what is heard by the listener. It serves as a simplification for psychoacoustic sound field synthesis for music presented in this book. Finally, the chapter illustrates strategies to visualize measured sound radiation properties.

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This paper presents a novel bird monitoring and recognition system in noisy environments. The project objective is to avoid bird strikes to aircraft. First, a cost-effective microphone dish concept (microphone array with many concentric rings) is presented that can provide directional and accurate acquisition of bird sounds and can simultaneously pick up bird sounds from different directions. Second, direction-of-arrival (DOA) and beamforming algorithms have been developed for the circular array. Third, an efficient recognition algorithm is proposed which uses Gaussian mixture models (GMMs). The overall system is suitable for monitoring and recognition for a large number of birds. Fourth, a hardware prototype has been built and initial experiments demonstrated that the array can acquire and classify birds accurately.

This work presents a novel technique that performs both orientation and distance localization of a sound source in a three-dimensional (3D) space using only the interaural time difference (ITD) cue, generated by a newly-developed self-rotational bi-microphone robotic platform. The system dynamics is established in the spherical coordinate frame using a state-space model. The observability analysis of the state-space model shows that the system is unobservable when the sound source is placed with elevation angles of 90 and 0 degree. The proposed method utilizes the difference between the azimuth estimates resulting from respectively the 3D and the two-dimensional models to check the zero-degree-elevation condition and further estimates the elevation angle using a polynomial curve fitting approach. Also, the proposed method is capable of detecting a 90-degree elevation by extracting the zero-ITD signal 'buried' in noise. Additionally, a distance localization is performed by first rotating the microphone array to face toward the sound source and then shifting the microphone perpendicular to the source-robot vector by a predefined distance of a fixed number of steps. The integrated rotational and translational motions of the microphone array provide a complete orientation and distance localization using only the ITD cue. A novel robotic platform using a self-rotational bi-microphone array was also developed for unmanned ground robots performing sound source localization. The proposed technique was first tested in simulation and was then verified on the newly-developed robotic platform. Experimental data collected by the microphones installed on a KEMAR dummy head were also used to test the proposed technique. All results show the effectiveness of the proposed technique.

To date electric pianos and samplers tend to concentrate on authenticity in terms of temporal and spectral aspects of sound. However, they barely recreate the original sound radiation characteristics, which contribute to the perception of width and depth, vividness and voice separation, especially for instrumentalists, who are located near the instrument. To achieve this, a number of sound field measurement and synthesis techniques need to be applied and adequately combined. In this paper we present the theoretic foundation to combine so far isolated and fragmented sound field analysis and synthesis methods to realize a radiation keyboard, an electric harpsichord that approximates the sound of a real harpsichord precisely in time, frequency, and space domain. Potential applications for such a radiation keyboard are conservation of historic musical instruments, music performance, and psychoacoustic measurements for instrument and synthesizer building and for studies of music perception, cognition, and embodiment.

In this paper we propose a methodology for the localization of virtual acoustic sources for sound field rendering applications. After the reconstruction of the sound field in the listening area by means of circular harmonic decomposition, the virtual source location is found through the Hough transform. We prove the accuracy of the proposed methodology by comparing the source locations estimates
with those of a subjective test campaign.

A new aeroacoustic measurement capability has been developed for use in open-jet testing in the NASA Langley 14- by 22-Foot Subsonic Tunnel (14x22 tunnel). A suite of instruments has been developed to characterize noise source strengths, locations, and directivity for both semi-span and full-span test articles in the facility. The primary instrument of the suite is a fully traversable microphone phased array for identification of noise source locations and strengths on models. The array can be mounted in the ceiling or on either side of the facility test section to accommodate various test article configurations. Complementing the phased array is an ensemble of streamwise traversing microphones that can be placed around the test section at defined locations to conduct noise source directivity studies along both flyover and sideline axes. A customized data acquisition system has been developed for the instrumentation suite that allows for command and control of all aspects of the array and microphone hardware, and is coupled with a comprehensive data reduction system to generate information in near real time. This information includes such items as time histories and spectral data for individual microphones and groups of microphones, contour presentations of noise source locations and strengths, and hemispherical directivity data. The data acquisition system integrates with the 14x22 tunnel data system to allow real time capture of facility parameters during acquisition of microphone data. Performance analysis of the array using a coarray figure of merit and synthetic point spread functions using monopole sources show that the array design is adequate for its intended purpose in the facility. Although the instrumentation suite is designed to characterize noise for a variety of test articles in the 14x22 tunnel, this paper concentrates on use of the instruments for two specific campaigns in the facility, namely a full-span NASA Hybrid Wing Body (HWB) model entry and a semi-span Gulfstream aircraft model entry, tested in the facility in the winter of 2012 and spring of 2013, respectively.

The general spiral equation is an excellent tool in array development. However, some modifications can improve the array response for a certain range of frequencies. This work describes a different approach to the spiral equation aiming the project of a high-frequency array (20 k - 35 kHz). By constraining some of the array and measurements parameters, a geometry is proposed intending to maximize the dynamic range response at a focal point. The steps of this approach are discussed showing the results of the simulations for different parameters of this equation. Some of these parameters were fixed and for some others a parameter sweep was performed. The final geometry is intended to work with ultrasound applications, such as measurements of scale models, and also with the acoustic frequency range.

—This paper describes results of the research on enhancement of loudspeaker impulse response measurement in small room. Proposed enhancement is a combination of MLS (Maximum Length Sequence) algorithm and Delay-Sum Beamforming in order to measure frequency response of loudspeakers even in very small and poorly suppressed rooms. Application of the beamforming algorithm allows the extension measure of impulse response by damping reflections from walls, floor and ceiling. Streszczenie – w artykule opisano wyniki badań dotyczą-cych poprawy jakości pomiaru odpowiedzi impulsowej gło-śników dokonywanego w małym pomieszczeniu. Propono-wana metoda poprawy jakości pomiaru jest połączeniem algorytmu MLS (ang. maximum-length sequence) oraz kształtowania wiązki na zasadzie sumowania opóźnionych sygnałów (ang. delay-sum beamforming) w celu pomiaru charakterystyki amplitudowo-częstotliwościowej głośników nawet w bardzo małych i słabo wyciszonych pomieszcze-niach. Zastosowanie algorytmu kształtowania wiązki umoż-liwia pomiar odpowiedzi impulsowej poprzez wytłumienie odbić od ścian, podłogi i sufitu.

In this paper we propose a novel fast algorithm for calculating the transfer function of the Zelinski post-filter in a microphone array. The proposed algorithm requires less memory and fewer arithmetical multiplications. We demonstrate that for the " classical " algorithm computational complexity increases quadratically as a function of the number of microphones in the array. In contrast, the computational complexity of the proposed algorithm increases linearly. This provides a considerable acceleration in the calculation of the post-filter transfer function in real-time systems.

The paper presents improvements to the methodology for the direct estimation of the spatiospectral distribution of an acoustic source from microphone measurements that comprise fixed and continuously scanning sensors. The signals from the scanning sensors are non-stationary due to the time-varying source-sensor distance and the traversing of an acoustic field with spatially varying statistics. Quasi-stationarity is sought by dividing the signals into blocks and applying of a frequency-dependent window within each block. The effects of the block distribution and width of the window on the acoustic source images are analyzed and guidelines are developed for optimal block sizes. The methodology is applied to acoustic fields emitted by a subsonic jet in isolation and surrounded by a plate in shielding and reflection orientations. The experimental setup consists of a microphone phased array with one scanning and multiple fixed sensors. The ability of the continuous-scan paradigm, coupled with the improved signal division, to provide high-definition noise source maps is demonstrated.

The use of microphone arrays for sound-source localization is a well-researched topic. The response of such sensor arrays is dependent on the quantity of microphones operating on the array. A higher number of microphones, however, increase the computational demand, making real-time response challenging. In this paper, we present a Filter-and-Sum based architecture and several acceleration techniques to provide accurate sound-source localization in real-time. Experiments demonstrate how an accurate sound-source localization is obtained in a couple of milliseconds, independently of the number of microphones. Finally, we also propose different strategies to further accelerate the sound-source localization while offering increased angular resolution.

In all speech communication settings the quality and intelligibility of speech is of utmost importance for ease and accuracy of information exchange. The Spectral Subtraction Technique is one of the methods to reduce stationary noise but the non stationary noise still passes through it. Spectral Subtraction also introduces a musical noise which is very annoying to human ears. Beamforming is another possible method of speech enhancement that can be used. Beamforming by itself, however, does not appear to provide enough improvement. Further, the performance of Beamforming becomes worse if the noise source comes from many directions or the speech has strong reverberation. A combined technique using the advantages of Spectral Subtraction and Beamforming Techniques is proposed where the Spectral Subtraction Technique followed by Beamforming Technique reduces stationary as well as residual, musical noise. It can be observed that the Spectral Subtraction followed by Beamforming gives better SNR value as compared to that of individual techniques, thereby improving the quality of speech. Numerous simulation results are used to illustrate the reasoning.

This paper describes preliminary research results on enhancement of loudspeaker impulse response measurement in small room. Proposed enhancement is a combination of MLS (Maximum Length Sequence) algorithm and Delay-Sum Beamforming in order to measure frequency response of loudspeakers even in very small and poorly suppressed rooms. Application of the beamforming algorithm allows the extension measure of impulse response by damping reflections from walls, floor and ceiling.

This paper presents a novel three-dimensional (3D) sound source localization (SSL) technique based on only Interaural Time Difference (ITD) signals, acquired by a self-rotational two-microphone array on an Unmanned Ground Vehicle. Both the azimuth and elevation angles of a stationary sound source are identified using the phase angle and amplitude of the acquired ITD signal. An SSL algorithm based on an extended Kalman filter (EKF) is developed. The observability analysis reveals the singularity of the state when the sound source is placed above the microphone array. A means of detecting this singularity is then proposed and incorporated into the proposed SSL algorithm. The proposed technique is tested in both a simulated environment and two hardware platforms, i.e., a KEMAR dummy binaural head and a robotic platform. All results show the fast and accurate convergence of estimates.

In this paper we present the results of a comparative study of algorithms for speech signal processing in a microphone array. We compared the multichannel alignment method and two modifications of the well-known Zel-inski post-filtering. Comparisons were performed using artificial and real signals recorded in real noisy environments. The experiments helped us to devise recommendations for choosing a suitable method of signal processing for different noise conditions.

In this paper we propose a methodology aimed at improving the resolution capabilities of plenacoustic imaging, which is based on deconvolution techniques mutuated from aerospace acoustic imaging. In order to reduce the computational burden, we also propose a modification of the minimization problem that exploits the highly structured information contained in the plenacoustic image. Experiments and simulations show the improvement of the accuracy gained by applying the deconvolution operator.

In this paper, a new learning approach for sound source localization is presented using ad hoc either synchronous or asynchronous distributed microphone networks based on time differences of arrival (TDOA) estimation. It is first to propose a new concept in which coordinates of a sound source location are defined as functions of TDOAs, computing for each pair of microphone signals in the network. Then, given a set of pre-recorded sound measurements and their corresponding source locations, the multilevel B-Splines based learning model is proposed to be trained by the input of the known TDOAs and the output of the known coordinates of the sound source locations. For a new acoustic source, if its sound signals are recorded, the correspondingly computed TDOAs can be fed into the learned model to predict the location of the new
source. Superiorities of the proposed method are to incorporate acoustic characteristics of a targeted environment and even remaining uncertainty of TDOA estimations into the learning model before conducting its prediction and to be applicable for both synchronous or asynchronous distributed microphone sensor networks. Effectiveness of the proposed algorithm in terms of localization accuracy and computational cost in comparisons with state-of-the-art methods was extensively validated on both synthetic simulation experiments as well as in three real-life environments.

Localization of the sound source using microphone arrays is largely done by Acoustical Holography or Beamforming method. In this paper we present an implementation of Delay-And-Sum algorithm for beamforming computation using various microphone arrays. A MATLAB code was implemented for postprocessing of obtained acoustic signals. There is also included a description of the fractional delay filter implementation to delay the signal samples by racional values of sample period. The goal of this paper was comparing the accuracy of the source localization with the microphone arrays using small number of microphones. The results of the localization of several noise sources by the different types of arrays is presented.

This paper presents a new adaptive technique for speech capturing in adverse conditions using microphone arrays. The proposed technique is based on frequency-domain alignment of microphone signals with the output of the fixed beamformer directed to the target speaker. This alignment procedure improves pattern directivity and reduces sidelobes. The low complexity of the technique is achieved by means of a frequency-domain implementation of the algorithm. This makes it possible to implement this technique in real-time applications with a large number of microphones. The technique was evaluated on speech data corrupted by varying levels and directions of noise and interference. The proposed technique improves the directivity pattern of a traditional Dealy & Sum beamformer as well as provides additional suppression of spatially incoherent noise, diffuse noise and interference, with minimal loss of the target signal quality.

Electromagnetic articulography (EMA) is one of the instrumental phonetic research methods used for recording and assessing ar-ticulatory movements. Usually, articulographic data are analysed together with standard audio recordings. This paper, however, demonstrates how coupling the articulograph with devices providing other types of information may be used in more advanced speech research. A novel measurement system is presented that consists of the AG 500 electromagnetic articulograph, a 16-channel microphone array with a dedicated audio recorder and a video module consisting of 3 high-speed cameras. It is argued that synchronization of all these devices allows for comparative analyses of results obtained with the three components. To complement the description of the system, the article presents innovative data analysis techniques developed by the authors as well as preliminary results of the system's accuracy.

Speech enhancement aims to improve the speech quality by using
various techniques. Spectral Subtraction Technique is one earliest
and longer standing, popular approaches to noise compensation
and speech enhancement. It reduces stationary noise but the non
stationary noise still passes through it. Further, it also introduces a
musical noise which is very annoying to human ears. Beamforming
is another possible method of speech enhancement that can be
used. Beamforming by itself, however, does not appear to provide
enough improvement. Further, the performance of Beamforming
becomes worse if the noise source comes from many directions or
the speech has strong reverberation. A combined technique using
the Spectral Subtraction Technique followed by Beamforming
Technique reduces stationary as well as residual, musical noise. It
can be observed that the Spectral Subtraction followed by
Beamforming gives better SNR value as compared to that of
individual techniques, thereby improving the quality of speech.
Numerous simulation results are used to illustrate the reasoning.

Electromagnetic Articulography (EMA) is one of the methods of examining tongue movements. This technique is usually supported by audio recordings and may be supplemented by video data in order to analyze external articulator movements. In the research presented here, a new approach using a microphone array and synchronous video recordings is proposed and an examination of Polish nasal consonant is presented as an example of this novel method of analysis.

In this paper, we present three approaches to localizing and tracking a sound source moving in a three-dimensional (3D) space using a bi-microphone array rotating at a fixed angular velocity. The motion of the sound source along with the rotation of the bi-microphone array results in a sinusoidal inter-channel time difference (ICTD) signal with time-varying amplitude and phase. Two state-space models were employed to develop extended Kalman filters (EKFs) that identify instantaneous amplitude and phase of the signal. Observability analysis of the two state-space models was conducted to reveal singularities. We also developed a method based on Hilbert transform, which is done by comparing the analytic signal of the true ICTD signal with that of a virtual signal having zero elevation and azimuth angles. A moving average filter is then applied to reduce the noise and the effect of the artifacts at the beginning and the ending portion of the estimates. The effectiveness of the proposed methods was tested and comparison studies were conducted in the simulation.

– Ultrasonic components of bird voices are rarely analyzed in bioacoustical papers. The main reason behind this situation is an equipment used for digital recording of bird voices in which: 1) sampling frequencies do not usually exceed 48 kHz and 2) microphones that limits bandwidth of recorded bird voices are also commonly used. The experiments described in this paper were carried out with 96 kHz sampling frequency and high quality broadband microphones. Moreover, because the commonly used 1-bit sigma-delta A/D converters introduce some distortion to the signal – they particularly affect its high frequency components. In the research we propose using the SAR converter instead. Preliminary automatic recognition research comparing performance of SAR and 1-bit sigma-delta A/D is also presented in the paper. Investigation concerned analysis of several voices of passerine birds e.g. Parus major, Passer montanus in ultrasonic range which unveiled existence of strong ultrasonic tonal frequency components and formants in the analyzed bird voices. Methodology that allowed us to record bird voices in an ultrasound range was described. Additionally, the general conclusion outlining the best bird voices recording strategy has been drawn. Biological meaning of the ultrasound components in bird voices still remains an open research question.

System which can monitors bird species based on automatic bird voices recognition is very useful in order to protect biodiversity of avifauna. In the paper the electronic devices being integral part of the proposed acoustical bird monitoring system are described. These devices are: digital camera, digital camcorder, multi-channel digital audio recorder, digital alluring device, microphone array, GPS localizator, radio communication module and portable weather station. Some guidelines resulting from authors experience in using described equipment are also given.

Beamforming in reverberant environments is important to locate and quantify noise sources in turbofan engine nacelles, automobile interiors, factories, and architectural settings. In order to validate and explore the limits of this approach, a beamforming experiment was conducted in a reverberation chamber using a 32 channel planar phased array and B&K sound power source. In the reference configuration, the source was located 2 m from the 1.1 m diameter array, and neither was close to the chamber walls. A less-demanding case was constructed by adding some foam absorber to the room to reduce reverberation. A difficult case resulted when the source was placed 5 m from the array, about 1 m from a corner of the chamber. Conventional frequency-domain beamforming with diagonal deletion was applied. The sound source was accurately located at the 2 m distance, with and without the added absorber. In the 5 m case, the sound source could be located at only a few frequencies and only when the processing bandwidth was increased from 48.8 Hz to 781 Hz. Processing individual eigenvectors of the CSM separated the direct and reflected source in 5 m case. The error in the deduced broadband sound power was 0.72 dB in the baseline case, 1.47 dB with the added absorber, and 5.41 dB with the speaker in the corner. Application of CLEAN-SC did not improve the accuracy of the corner results. Use of a Green’s function that attempts to account for reflections was ineffective. It is concluded that beamforming in highly reverberant environment with the free space Green’s function is practical, provided the array is designed and positioned correctly for the environment and source location.

The shakuhachi, a Japanese flute, is a rather small instrument with a simple geometry.  Still, it appears to have acomplicated spatial sound radiation characteristic.  This effect results from interference of sound emanating fromfinger holes and the blowing hole as well as diffraction around and acoustic shadow behind the instrumentalist.Even in absence of room reflections, the pure direct sound of musical instruments already creates the impressionof a certain extent of the source. This perceived extent is especially large for listeners close to the instrument anddecreases with distance. This effect is investigated in more detail on the shakuhachi.The sound of a shakuhachi is recorded in an anechoic chamber by a circular microphone array consisting of 128microphones. Amplitude and phase per frequency and angle around the instrument are measured. Interaural phase-and amplitude differences as well as the correlation of the signals arriving at the two ears are calculated for severallistening  positions  at  various  angles  and  distances.  These  parameters  are  compared  between  different  playingtechniques. It is discussed how far the parameters are suitable to explain the perception of the spatial source extent.

Beamforming is an acoustic imaging technique that can estimate the radiation pattern of simple or complex sound sources and produce a map of the results. The pass-by noise test is a standardized test that aims to evaluate the overall noise of vehicle’s sideline. By coupling the idea of pass-by test with the extension of beamforming technique to moving sources provides the access to the recognition of sound sources produced by vehicle movements, for example, rolling tires, engines and exhaust systems. The present paper aims to describe a low cost system to apply the beamforming technique to pass-by noise test. The system is based on the use of low cost electret microphones mounted in a metallic array which are connected by a coaxial cable to the acquisition system. Later in this document in the application section the results of beamforming maps of pass-by noise test can be viewed in more detail.

This paper presents a new method of improving microphone array di-rectivity in the low-frequency domain. The method is based on a sub-band processing technique. We also evaluate the parameters and characteristics of the method and consider some of its practical implementations.

The sound radiation characteristic is a feature of instru- mental sound which contributes to the perception of a certain source extent or apparent source width (ASW) even under anechoic conditions. In a study physical parame- ters are derived or adopted from room acoustical inves- tigations, based on psychoacoustic considerations, which are assumed to be related to the ASW. Linear regression between some parameters and the physical source extent (width) exhibit a significant (p < 0.05) correlation. Multi- ple regressions with combinations of a monaural and a bin- aural parameter even reveal highly significant (p < 0.01) relations. As they correlate with the physical source extent, these parameters are assumed to be suitable for the inves- tigation of perceived source extent in a listening test with artificially created sound radiations under preparation.

Feedforward active noise control (ANC) system are widely used to reduce the wide-band noise in different application. In feedforward ANC systems, when the noise source is unknown, the misplacement of the reference microphone may violate the causality constraint. We present a performance analysis of the feedforward ANC system under a noncausal condition. The ANC system performance degrades when the degree of noncausality increases. This research applies the microphone array technique to feedforward ANC systems to solve the unknown noise source problem. The generalized cross-correlation (GCC) and steering response power (SRP) methods based on microphone array are used to estimate the noise source location. Then, the ANC system selects the proper reference microphone for a noise control algorithm. The simulation and experiment results show that the SRP method can estimate the noise source direction with 84% accuracy. The proposed microphone array integrated ANC system can dramatically improve the system performance.

In all speech communication settings, the quality and intelligibility of speech is of utmost importance for ease and accuracy of information exchange. The speech processing systems used to communicate or store speech are usually designed for a noise free environment but in a real-world environment, the presence of background interference in the form of additive background and channel noise drastically degrades the performance of these systems, causing inaccurate information exchange and listener fatigue.
The Spectral Subtraction Technique can be used to reduce stationary noise but the non stationary noise still passes through it. Spectral Subtraction also introduces a musical noise which is very annoying to human ears. Beamforming is another possible method of speech enhancement that can be used. Further, the musical noise of Spectral Subtraction can be reduced by Beamforming. Beamforming by itself, however, does not appear to provide enough improvement. Further, the performance of Beamforming becomes worse if the noise source comes from many directions or the speech has strong reverberation.
Therefore, a system has been designed with a combination of Spectral
Subtraction Technique followed by Beamforming Technique reducing stationary as well as residual, musical noise. Algorithms and associated software have been developed for 1) Spectral Subtraction 2)
Beamforming Technique and 3) Spectral Subtraction followed
by Beamforming Technique. The last developed technique results in getting a noise free speech free of musical noise and reverberation making the speech intelligible and of good quality.
Processing of the signal for Spectral Subtraction, Delay Sum Beamforming and the Combined Techniques, was carried out individually for three different experiments (with 3 , 6 and 10 microphones) and for 4 different cases with 3 different signals and
fourth a signal with Gaussian white Noise. The SNR in each case was calculated.

Classical Beamforming is a signal processing technique capable of creating acoustic images. The algorithm utilized is the simple delay-and-sum over the sampled sound waves in the time domain. One important tool in the application of the technique is the array (or antenna), which is composed of a set of microphones, organized in a specific geometry. Depending on its construction, the resulting acoustic images will have different patterns. For this reason, a research has to be performed on the array's specifications to and for the best performance depending on the application. The way to test the performance is to evaluate the PSFs (Point Spread Functions) of the geometries, which is dependent on the width of the analysis' bandwidth. Hence, this work will compare and discuss some aspects of bandwidth on PSFs simulations regarding four different array geometries.

Hybrid cars have become more important due to increasing fuel costs and the goal of reducing CO2 emissions. In this talk, the noise producing mechanisms of a Honda Civic Hybrid will be studied using a phased array and a range of beamforming techniques. The importance of multiple propagation paths, incoherent and coherent source distributions, tone and broadband noise, and array focus distance will be illustrated. Frequencies to be considered run from 100 Hz to 30 kHz. Methods to be discussed and compared include classical beamforming in 2D and 3D, beamforming with enhanced resolution, CLEAN-SC, and DAMAS2. Emphasis will be given to exterior sounds, with some treatment of the interior case. The goal of the presentation will be to give a basic understanding of beamforming techniques including software and hardware and to illustrate their wide applicability to common noise control problems and the implications of the choice of method for different types of noise sources.

Beamforming microphone antenna has been widely used to locate sound sources and build acoustic model. The particularities of this technique are the use of a relatively few number of  microphones and simplification in the signal processing. So, this technique is fast and easy to be carried out. But its main inconvenience is the poor resolution at low frequency. In spite of this, this technique has been applied in the fields of the pass-by noise analysis, the aero-acoustic noise identification and the source localization in cabins. However, for the case of cabins, the single layer antenna does not allow one to distinguish sources located on both sides of antenna, causing difficulties to explain the results.
In this paper, a double-layer microphone antenna (two layers of microphones or a layer of microphone and a layer of velocity sensors) applied in cabins is examined. This method uses the sound pressure and particle velocity of measurements together with a Statistically Optimal Array Processing (SOAP) to identify wave front of a source model. It is shown how a processing method similar to that of the sound intensity allows one to separate the contributions which come from two sides of antenna and are made by sources (leakage, radiating zones etc.) located on the walls of the cabin. Numerical simulations in a multi-sources environment demonstrates the interests of this technique.